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platform-external-webrtc/media/base
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Ilya Nikolaevskiy 3288168e4f Enable video adaptation for all screenshare content 
Since screenshare uses "Maintain resolution" degradation preference,
adapting should be enabled to reduce framerate if encoder can't keep up.

Bug: chromium:690537
Change-Id: I1f4418b7b7b4faa13f34d5353e3c625a59975c05
Reviewed-on: https://webrtc-review.googlesource.com/95460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24426}
2018-08-24 12:02:03 +00:00
..
adaptedvideotracksource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
adaptedvideotracksource.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
audiosource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
codec_unittest.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
device.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeframesource.cc
Refactor PeerConnectionIntegrationTest to not use cricket::VideoCapturer
2018-05-11 09:22:20 +00:00
fakeframesource.h
Refactor PeerConnectionIntegrationTest to not use cricket::VideoCapturer
2018-05-11 09:22:20 +00:00
fakemediaengine.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakenetworkinterface.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakevideocapturer.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakevideocapturer.h
Introduce TaskQueueForTest.
2018-02-14 15:32:49 +00:00
fakevideorenderer.cc
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
fakevideorenderer.h
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
h264_profile_level_id.cc
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
h264_profile_level_id.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
mediachannel.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
mediachannel.h
Delete unused member MediaSenderInfo::packets_cached.
2018-08-10 13:13:54 +00:00
mediaconfig.h
Remove wire-up for cancelled experement on VAAPI VP8 encoding
2018-06-15 10:04:07 +00:00
mediaconstants.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconstants.h
Add RRTR parameter to media engine and pass it to video receive stream
2018-04-06 11:15:12 +00:00
mediaengine.cc
Add Rtcp parameters for PeerConnection senders
2018-05-28 09:28:59 +00:00
mediaengine.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
turnutils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turnutils.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
turnutils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videoadapter_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
videoadapter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videoadapter.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
videobroadcaster_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
videocapturer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocapturer.cc
Enable video adaptation for all screenshare content
2018-08-24 12:02:03 +00:00
videocapturer.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videocapturerfactory.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videocommon_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videocommon.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocommon.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videosourcebase.cc
Enable and fix chromium clang warnings in sdk/android targets.
2018-04-05 11:22:03 +00:00
videosourcebase.h
Move VideoStreamEncoderInterface to api/.
2018-05-21 19:50:37 +00:00
vp9_profile.cc
Reland "Remove rtc::Optional alias and api:optional target"
2018-07-11 19:02:51 +00:00
vp9_profile.h
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00